Re: [asterisk-users] RTP audio
On Tue, Oct 18, 2022 at 4:56 PM Jerry Geis wrote: > > > On Tue, Oct 18, 2022 at 3:38 PM Jerry Geis wrote: > >> Has there been issues where "once in a while" RTP audio does not work ? >> >> Example: connection to Cisco call manager - works mostly all the time. >> >> once in a great while - person does not hear the "beep" when calling in. >> once in a great while - person they hear the beep - but do not hear the >> audio public address. >> >> What would I be looking for to track this beast down ? >> >> This is my SIP trunk >> [LSVOIP] >> type=friend >> dtmfmode=rfc2833 >> secret=password >> username=LSVOIP >> defaultuser=LSVOIP >> disallow=all >> allow=ulaw >> allow=alaw >> context=incoming >> host=172.1.1.1 >> canreinvite=yes >> qualify=yes >> insecure=invite >> >> Thoughts? >> >> Jerry >> > > > Is there any kind of pjsip vs old SIP (which I am using) issue happening > here. (asterisk 18.14.0) > No. The media stack between the two is the same, and is the existing one that has existed for years. The starting point for any issue like this is a packet capture that you can examine in wireshark to see what media is flowing, if any, where, and the signaling. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoIP support engineer opportunity
Hello, Voisonics is hiring a VoIP support engineer to assist our customers running Asterisk based hosted PBX platforms. This is a part-time contract work-from-home position. For communication reasons we're looking for someone in a timezone encompassing New Zealand, Canada, the USA, and Mexico. If you are not physically located in that area please do not apply - being "flexible" from another part of the world is not what we're looking for. The role involves providing technical support of Asterisk based PBX platforms to our customer's technical staff, Linux system administration, and small dev-ops type development projects. It does not involve providing technical support to end users or the general public. Customers are located around the world. You will generally be responding during your business hours, though sometimes out of hours work will be necessary. Once training is completed, the position will involve providing 24x7 on-call emergency cover in rotation with other staff. Must-have: 1. Fluent command-line Linux ability on Ubuntu, Debian, Rocky, CentOS, and/or RedHat. 2. Asterisk administration and configuration experience. 3. SIP debugging experience. For example, you should know what packets are typically involved in setting up a call. 4. Experience with administration and configuration of Apache and MySQL or PostgreSQL. 5. Ability to program in Perl or shell scripts. 6. Good written and verbal English language ability. 7. Ability to solve problems and create solutions independently. Although there are other staff, most work is done by one engineer on their own. 8. Have experience providing professional IT support to business. 9. Be an individual self-employed contractor. Nice-to-have: 1. Experience with Kamailio, NFS, GlusterFS, Puppet, or Zabbix. 2. C, Go, Javascript, AngularJS, HTML or CSS programming ability. 3. Advanced network knowledge (beyond basic Linux networking which is a must-have). To apply for this role please email me off-list. Do not call by phone unless arranged in advance. In your application: 1. List your experience/compatibility for each of the must-have requirements individually, plus any of the nice-to-have items you fit as well. 2. Provide your physical location, hours of availability, and indication of hourly rate. 3. Let us know what other work you already have during your hours of availability. 4. A full CV is welcome. Thank you, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP audio
On Tue, Oct 18, 2022 at 3:38 PM Jerry Geis wrote: > Has there been issues where "once in a while" RTP audio does not work ? > > Example: connection to Cisco call manager - works mostly all the time. > > once in a great while - person does not hear the "beep" when calling in. > once in a great while - person they hear the beep - but do not hear the > audio public address. > > What would I be looking for to track this beast down ? > > This is my SIP trunk > [LSVOIP] > type=friend > dtmfmode=rfc2833 > secret=password > username=LSVOIP > defaultuser=LSVOIP > disallow=all > allow=ulaw > allow=alaw > context=incoming > host=172.1.1.1 > canreinvite=yes > qualify=yes > insecure=invite > > Thoughts? > > Jerry > Is there any kind of pjsip vs old SIP (which I am using) issue happening here. (asterisk 18.14.0) Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP audio
Has there been issues where "once in a while" RTP audio does not work ? Example: connection to Cisco call manager - works mostly all the time. once in a great while - person does not hear the "beep" when calling in. once in a great while - person they hear the beep - but do not hear the audio public address. What would I be looking for to track this beast down ? This is my SIP trunk [LSVOIP] type=friend dtmfmode=rfc2833 secret=password username=LSVOIP defaultuser=LSVOIP disallow=all allow=ulaw allow=alaw context=incoming host=172.1.1.1 canreinvite=yes qualify=yes insecure=invite Thoughts? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users